[asterisk-users] random one way audio and noise between SIP phoneson same LAN

[Giorgio Incantalupo]

Hi Scott,
seems that we have the same problem...I have canreinvite=no and polycom 
phones.
I do not have cisco switch and qualify=yes but I think that is not the 
problem.

I've got 2 questions:
1) my polycom firmware is:
sip.ver: 1.6.5.0043
bootrom.ver: 2_6_2

what are yours?
2) have you got one way calls only or noise on sip calls conversations too?

TIA


Giorgio Incantalupo

P.S.: for configuration/monitoring apps  I'm still on it...I hope to 
find useful tools asap. In case, I'll let you know.


Scott Scecina wrote:
> I'm having the same "random" problem.
>
> I have "canreinvite=no" on all extensions.  I have "qualify => yes" on all
> non-NAT extensions. I do have several NAT extensions, but I've not had
> reports of problems from those.  95% of my extensions (all polycom 501/601)
> are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches.
>
> In all cases, the called party cannot hear the calling party.  The calling
> party has the "still ringing" icon on their phone, but can hear the called
> party talking.  I've got call monitoring turned on, and asterisk is recording
> both sides of the conversation.   
>
> The problem occurs on SIP->SIP and Zap->SIP calls. 
>
> I've tried enabling sip debug on a particular extension that seemed to be
> experiencing the problem more than others.  However the problem did not occur
> when the debugging was on.
>
> Sip debug generates so much noise I've been hesitant to turn it on
> system-wide.  Is there a way I can turn on sip debug and have all that
> logging go to a specific file (and not in the asterisk console)?
>
> Also, are there any other configuration/logging tricks I can try?
>
> Thank you,
>
> Scott Scecina
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Klaus Darilion
> Sent: Wednesday, October 18, 2006 8:48 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] random one way audio and noise between SIP
> phoneson same LAN
>
> Do you use canreinvite (sip.conf)?
>
> Change the setting (setting canreinvite=yes may cause nat problems) nad 
> verify if the problem still appears.
>
> Using htis setting you can find out if the Audio problem occurs only 
> when media is relayed via Asterisk (->the problem is caused by Asterisk) 
> or in all cases (the problem is not caused by Asterisk)
>
> regards
> klaus
>
> Giorgio Incantalupo wrote:
>   
>> Hi,
>> sometimes I have one way calls and noise between sip phones connected to 
>> the same LAN so no nat/firewall is involved. I tried with different sip 
>> phone models soft phones and the result is the same. I searched inside 
>> every log file but found nothing. I made different PBX with different 
>> hardware but the result is always the same.
>>
>> Is there anybody experiencing this terrible problem?
>> Considering to monitor a remote PBX via ssh, which  text-only 
>> application  could  I use?
>>
>> TIA
>>
>> Giorgio Incantalupo
>>
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